[asterisk-users] Help with voicemail

[Eric Germann]

I'm looking to use * for a HQ/branch office topology with fairly few calls
over the WAN.  The questions I have all pertain to the following
architectural pic:  http://www.45891.com/misc/arch.jpg

I'm looking at a distributed architecture so users are somewhat functional
when the link to HQ is down, with a centralized voicemail server to allow
for transfer of voicemail messages from user to user, on both the VoIP and
legacy system (voicemail being on a dedicated * box).

1.	Thanks to jporier who can be found at ccu.edu, I figured out how to
deal with MWI for all the remote servers by mounting the voicemail directory
via NFS from VMAIL1 onto the VOIPx servers which host the actual phones.
Then sticking a msg00000.txt file into the directory makes the blinky light
go on the phones.  So far so good.

What I'm asking the list for is either a brief code snippet or pointers to a
doc/link on how to setup the following:

A.	None of the VOIPx servers have vmail enabled on them.  When someone
gets dumped to voicemail, I envision the call being transferred to the
VMAIL1 server and it routing it directly to a mailbox for the user.

B.	VMAIL1 has no user extensions on it, just mailboxes.  It gets a call
on the trunk and dumps it to the appropriate vmail box based on the
extension that was called.

C.	How do I force the vmail to go down the trunk to VMAIL1?

D.	How do I catch it on the other end and stick it only in a mailbox?

Basically, how do I split the voicemail transfer off the local box to
another?


Now for a couple of architectural questions:

1.	When a caller rings thru the TANDEM1 box to a VOIP1 extension, and
then gets dumped to vmail, does the call go TANDEM1<->VOIP1<->VMAIL1 or does
VOIP1 hand it off so it's only TANDEM1<->VMAIL1, presuming all IAX2 trunks
are running a matching subset of codecs?

2.	Same thing for intracompany calls.  If VOIP2 calls VOIP1 user via
the tandem and gets dumped to vmail, does it go VOIP2<->VOIP1<->VMAIL1 or
VOIP2<->VMAIL1?  When user is talking on PSTN over Teliax, I can see TANDEM1
doing the transcoding if necessary and bridging via "IAX2 show peers".  This
leads me to believe it would go the former route, not the latter.  If it is
the former, is there a way to "make it" do the latter?

3.	For the TANDEM1 to VMAIL1 trunk, does it make sense to do G711 as
well on the trunk so it can transfer without transcoding to the voicemail
box (user dials the "voicemail number" DID on PRI from Embarq, hits the
mapping on the tandem and goes down the VMAIL1 trunk).

4.	Does it make sense to have a redundant tandem running on another box
and split the PRI's from the IAX trunks?  Embarq is looking into forwarding
the PRI DID blocks to the pilot number for the IAX2 trunk from Teliax so
when it goes down or is all-trunks-busy, it comes down the 'Net pipe.  Nice
to have Embarq on one side of the road ariel and TW underground on the other
side with separate entrances.

5.	When a call is hairpinned in TANDEM1 from the Embarq PRI to the tie
PRI's, is there any CPU overhead involved or is it basically done in the
card, presuming matching codecs on the PRI's?  Card is a digium TE405P quad
PRI card. 


Some implementation notes:

1.	All the boxes with IP addresses shown in the pic are setup.  I have
successful calls going Teliax -> Tandem -> VOIP1 and also back out to the
PSTN via the Tandem.  VOIP2 comes up tomorrow.  PRI's are a middle of the
night job later this week.
2.	All are running Trixbox 2.0b2.
3.	We're playing with codecs to see what gives the best quality for the
bandwidth.  Voip-info.org seems to point towards ilbc as having the lowest
overhead, followed by gsm and g729.  I presume if we want to bring fax in
off the Embarq PRI and/or Teliax we're going to have to use G.711u thru to
the Hylafax server with iaxmodem.  Anybody have any experience with bringing
fax in over a IAX2 trunk from Teliax (or any other voip provider for that
matter)?  We're switching this Thursday to a 10Mbps symmetric fiber
connection from Time Warner Business Class.

Once I get this working, I'm willing to write up a how-to (I'm going to have
to anyways for documentation, just needs to be sanitized) and put a pointer
or the doc on voip-info.org

Thanks in advance.

EKG